The present invention concerns an apparatus for encoding/decoding an analog signal and method thereof, wherein an analog signal such as an audio and video signal is compressed to reduce the transferred data stored in a memory.
The Pulse Code Modulation (PCM) which, is typically known as a method of encoding a waveform, employs a communications system comprising a sampler, an analog-to-digital(A/D) converter of a transmitter and a digital-to-analog(A/D) converter of a receiver. The A/D converter comprises a quantizer and an encoder, while the D/A converter comprises a quantizer and a decoder. In such PCM communications system, a waveform is modulated as follows:
An input analog signal is firstly sampled by the sampler, and the sampled portions are quantized. Namely, the signal of a limited frequency band which has no frequency components exceeding the maximum frequency fm(Hz) is sampled at the sampling interval T(Nyquist sampling interval 1/2 fm sec).
The sampled signal is divided into several steps quantized to a constant value at the center of each step. The quantized signal at each step has a quantized level approximating the original signal. Thus the pulses are quantized according to their sizes encoded by the encoder into the effective combination of the pulses for each sample.
The signal which is encoded into digital is separated from the noise added in the transference, and detected by the quantizer for the existence of the pulses in each pulse interval, where two voltage level differences are compared so as to transfer the result as a series of the reproduced pulses to the decoder. In contrast to the encoder, the D/A converter of the decoder produces the quantized sample pulses array of multiple levels, which is filtered to remove the frequency components except the base band providing a reproduced waveform as shown in FIG. 1.
Such PCM communication system requires a plurality of pulses for transmitting a sampled signal and thus a greater amount of data than the other pulse modulation systems in transmitting the same data. This system most approximately reproduces the original signal, but requires the data according to the number of the bits to quantize the value of each sampling, thus increasing the amount of data. Namely, 64 Kbits per second are needed in order to quantize a speech signal in 8 bits by sampling at every 8 KHz.
In view of the above fact, there has been proposed an adaptive delta Modulation (ADM) in order to reduce the amount of the data within the range not to considerably distort the original signal. This modulation system adaptively varies the step size of the reproduced waveform according to the variation of the original waveform, so that the required amount of the step size is added or subtracted in order to reconstruct the original signal waveform from the signal waveform obtained by ADM. Namely, the step size is increased by the base step size when the step direction of "k" th clock waveform edge (where the waveform changes to "high" or "low") is the same as in the edge(k-1), reduced by the basic stepsize when the step direction is opposite, and maintains the base step in the direction of approximating the original signal, as shown in FIG. 2.
This ADM system makes it possible to compress the data by 1 bit quantization with sufficiently short sampling period because the samplings are very closely interrelated. Hence, there is needed 1 bit per 1 sample, so that 32 k samplings require 32 k bits per second, thereby reducing the amount of the data to a half of that required for the PCM system.
However, this system is not accommodated to the abrupt changes of the original signal so as to produce the slope-over load distortion and granular noise which causes the reproduced waveform to repeat 0,1,0,1 even in a slight change of the original signal, thus making it difficult to reproduce the original signal waveform.